mirror of
https://github.com/Monibuca/plugin-webrtc.git
synced 2025-09-27 03:06:29 +08:00
298 lines
8.5 KiB
Go
298 lines
8.5 KiB
Go
package webrtc
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import (
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"fmt"
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"io"
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"net"
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"net/http"
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"regexp"
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"strings"
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"time"
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"go.uber.org/zap"
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"m7s.live/engine/v4"
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_ "embed"
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"github.com/pion/interceptor"
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. "github.com/pion/webrtc/v4"
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"m7s.live/engine/v4/config"
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"m7s.live/engine/v4/util"
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"m7s.live/plugin/webrtc/v4/webrtc"
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)
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// }{[]string{
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// "stun:stun.ekiga.net",
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// "stun:stun.ideasip.com",
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// "stun:stun.schlund.de",
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// "stun:stun.stunprotocol.org:3478",
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// "stun:stun.voiparound.com",
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// "stun:stun.voipbuster.com",
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// "stun:stun.voipstunt.com",
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// "stun:stun.voxgratia.org",
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// "stun:stun.services.mozilla.com",
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// "stun:stun.xten.com",
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// "stun:stun.softjoys.com",
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// "stun:stunserver.org",
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// "stun:stun.schlund.de",
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// "stun:stun.rixtelecom.se",
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// "stun:stun.iptel.org",
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// "stun:stun.ideasip.com",
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// "stun:stun.fwdnet.net",
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// "stun:stun.ekiga.net",
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// "stun:stun01.sipphone.com",
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// }}
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// type udpConn struct {
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// conn *net.UDPConn
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// port int
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// }
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//go:embed publish.html
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var publishHTML []byte
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//go:embed subscribe.html
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var subscribeHTML []byte
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var (
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reg_level = regexp.MustCompile("profile-level-id=(4.+f)")
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)
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type WebRTCConfig struct {
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config.Publish
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config.Subscribe
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ICEServers []ICEServer
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PublicIP []string
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Port string `default:"tcp:9000"`
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PLI time.Duration `default:"2s"` // 视频流丢包后，发送PLI请求
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EnableOpus bool `default:"true"` // 是否启用opus编码
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EnableAv1 bool `default:"true"` // 是否启用av1编码
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m MediaEngine
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s SettingEngine
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api *API
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}
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func (conf *WebRTCConfig) OnEvent(event any) {
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switch event.(type) {
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case engine.FirstConfig:
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if len(conf.ICEServers) > 0 {
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for i := range conf.ICEServers {
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b, _ := conf.ICEServers[i].MarshalJSON()
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conf.ICEServers[i].UnmarshalJSON(b)
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}
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}
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webrtc.RegisterCodecs(&conf.m)
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if conf.EnableOpus {
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conf.m.RegisterCodec(RTPCodecParameters{
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RTPCodecCapability: RTPCodecCapability{MimeTypeOpus, 48000, 2, "minptime=10;useinbandfec=1", nil},
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PayloadType: 111,
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}, RTPCodecTypeAudio)
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}
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if conf.EnableAv1 {
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conf.m.RegisterCodec(RTPCodecParameters{
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RTPCodecCapability: RTPCodecCapability{MimeTypeAV1, 90000, 0, "profile=2;level-idx=8;tier=1", nil},
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PayloadType: 45,
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}, RTPCodecTypeVideo)
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}
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i := &interceptor.Registry{}
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if len(conf.PublicIP) > 0 {
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conf.s.SetNAT1To1IPs(conf.PublicIP, ICECandidateTypeHost)
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}
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protocol, ports := util.Conf2Listener(conf.Port)
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if len(ports) == 0 {
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WebRTCPlugin.Fatal("webrtc port config error")
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}
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if protocol == "tcp" {
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tcpport := int(ports[0])
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tcpl, err := net.ListenTCP("tcp", &net.TCPAddr{
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IP: net.IP{0, 0, 0, 0},
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Port: tcpport,
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})
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if err != nil {
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WebRTCPlugin.Fatal("webrtc listener tcp", zap.Error(err))
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}
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WebRTCPlugin.Info("webrtc start listen", zap.Int("port", tcpport))
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conf.s.SetICETCPMux(NewICETCPMux(nil, tcpl, 4096))
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conf.s.SetNetworkTypes([]NetworkType{NetworkTypeTCP4, NetworkTypeTCP6})
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} else if len(ports) == 2 {
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conf.s.SetEphemeralUDPPortRange(ports[0], ports[1])
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} else {
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// 创建共享WEBRTC端口 默认9000
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udpListener, err := net.ListenUDP("udp", &net.UDPAddr{
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IP: net.IP{0, 0, 0, 0},
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Port: int(ports[0]),
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})
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if err != nil {
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WebRTCPlugin.Fatal("webrtc listener udp", zap.Error(err))
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}
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WebRTCPlugin.Info("webrtc start listen", zap.Uint16("port", ports[0]))
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conf.s.SetICEUDPMux(NewICEUDPMux(nil, udpListener))
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conf.s.SetNetworkTypes([]NetworkType{NetworkTypeUDP4, NetworkTypeUDP6})
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}
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if err := RegisterDefaultInterceptors(&conf.m, i); err != nil {
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panic(err)
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}
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conf.api = NewAPI(WithMediaEngine(&conf.m),
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WithInterceptorRegistry(i), WithSettingEngine(conf.s))
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}
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}
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func (conf *WebRTCConfig) Play_(w http.ResponseWriter, r *http.Request) {
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w.Header().Set("Content-Type", "application/sdp")
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streamPath := r.URL.Path[len("/play/"):]
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bytes, err := io.ReadAll(r.Body)
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var suber WebRTCSubscriber
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suber.SDP = string(bytes)
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suber.RemoteAddr = r.RemoteAddr
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if suber.PeerConnection, err = conf.api.NewPeerConnection(Configuration{
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ICEServers: conf.ICEServers,
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}); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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suber.OnICECandidate(func(ice *ICECandidate) {
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if ice != nil {
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suber.Info(ice.ToJSON().Candidate)
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}
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})
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if err = suber.SetRemoteDescription(SessionDescription{Type: SDPTypeOffer, SDP: suber.SDP}); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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if err = WebRTCPlugin.Subscribe(streamPath, &suber); err != nil {
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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if sdp, err := suber.GetAnswer(); err == nil {
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w.Write([]byte(sdp))
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} else {
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http.Error(w, err.Error(), http.StatusBadRequest)
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}
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}
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// https://datatracker.ietf.org/doc/html/draft-ietf-wish-whip
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func (conf *WebRTCConfig) Push_(w http.ResponseWriter, r *http.Request) {
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streamPath := r.URL.Path[len("/push/"):]
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rawQuery := r.URL.RawQuery
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auth := r.Header.Get("Authorization")
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if strings.HasPrefix(auth, "Bearer ") {
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auth = auth[len("Bearer "):]
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if rawQuery != "" {
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rawQuery += "&bearer=" + auth
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} else {
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rawQuery = "bearer=" + auth
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}
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WebRTCPlugin.Info("push", zap.String("stream", streamPath), zap.String("bearer", auth))
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}
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w.Header().Set("Content-Type", "application/sdp")
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w.Header().Set("Location", "/webrtc/api/stop/push/"+streamPath)
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if rawQuery != "" {
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streamPath += "?" + rawQuery
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}
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bytes, err := io.ReadAll(r.Body)
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var puber WebRTCPublisher
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puber.SDP = string(bytes)
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if puber.PeerConnection, err = conf.api.NewPeerConnection(Configuration{
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ICEServers: conf.ICEServers,
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}); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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puber.SetIO(puber.PeerConnection) //TODO: 单PC需要注释掉
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puber.OnICECandidate(func(ice *ICECandidate) {
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if ice != nil {
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puber.Info(ice.ToJSON().Candidate)
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}
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})
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puber.OnDataChannel(func(d *DataChannel) {
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puber.Info("OnDataChannel", zap.String("label", d.Label()))
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d.OnMessage(func(msg DataChannelMessage) {
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puber.SDP = string(msg.Data[1:])
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puber.Debug("dc message", zap.String("sdp", puber.SDP))
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if err := puber.SetRemoteDescription(SessionDescription{Type: SDPTypeOffer, SDP: puber.SDP}); err != nil {
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return
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}
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if answer, err := puber.GetAnswer(); err == nil {
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d.SendText(answer)
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} else {
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return
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}
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switch msg.Data[0] {
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case '0':
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puber.Stop()
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case '1':
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}
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})
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})
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// if _, err = puber.AddTransceiverFromKind(RTPCodecTypeVideo); err != nil {
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// http.Error(w, err.Error(), http.StatusInternalServerError)
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// return
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// }
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// if _, err = puber.AddTransceiverFromKind(RTPCodecTypeAudio); err != nil {
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// http.Error(w, err.Error(), http.StatusInternalServerError)
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// return
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// }
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if err = WebRTCPlugin.Publish(streamPath, &puber); err != nil {
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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puber.OnConnectionStateChange(func(state PeerConnectionState) {
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puber.Info("Connection State has changed:" + state.String())
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switch state {
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case PeerConnectionStateConnected:
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case PeerConnectionStateDisconnected, PeerConnectionStateFailed:
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puber.Stop()
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}
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})
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if err := puber.SetRemoteDescription(SessionDescription{Type: SDPTypeOffer, SDP: puber.SDP}); err != nil {
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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if answer, err := puber.GetAnswer(); err == nil {
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w.WriteHeader(http.StatusCreated)
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fmt.Fprint(w, answer)
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} else {
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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}
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func (conf *WebRTCConfig) Test_Publish(w http.ResponseWriter, r *http.Request) {
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w.Write(publishHTML)
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}
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func (conf *WebRTCConfig) Test_ScreenShare(w http.ResponseWriter, r *http.Request) {
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w.Write(publishHTML)
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}
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func (conf *WebRTCConfig) Test_Subscribe(w http.ResponseWriter, r *http.Request) {
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w.Write(subscribeHTML)
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}
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var webrtcConfig WebRTCConfig
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var WebRTCPlugin = engine.InstallPlugin(&webrtcConfig)
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func (conf *WebRTCConfig) Batch(w http.ResponseWriter, r *http.Request) {
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w.Header().Set("Content-Type", "application/sdp")
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bytes, err := io.ReadAll(r.Body)
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var suber WebRTCBatcher
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suber.RemoteAddr = r.RemoteAddr
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suber.SDP = string(bytes)
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if suber.PeerConnection, err = conf.api.NewPeerConnection(Configuration{
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ICEServers: conf.ICEServers,
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}); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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if err = suber.Start(); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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if sdp, err := suber.GetAnswer(); err == nil {
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w.Write([]byte(sdp))
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} else {
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http.Error(w, err.Error(), http.StatusBadRequest)
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}
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}
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